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Re: [Freeswitch-users] 2 voicemail questions
by Mark Crane 07 Jul '09

07 Jul '09

1. Can I email the voicemail message to multiple email addresses? If so, what format is this in? <param name="vm-mailto" value="me(a)myemail.com"/> I've been doing this successfully for quite some time using the mailer script that I wrote I just updated the wiki so that it would show the version I have been using that allows you to send multiple emails: You can use a comma or semi-colon between emails and send as many as you want. example: <param name="vm-mailto" value="me1@myemail.com;me2@myemail.com;me3@myemail.com"/> or <param name="vm-mailto" value="me1@myemail.com,me2@myemail.com,me3@myemail.com"/> http://wiki.freeswitch.org/wiki/PHP_email#mailer_app.php The script can send to a mail server to send over plain smtp, smtp authentication or even smtp tls which works with gmail. Mark J Crane mctch(a)yahoo.com --- On Tue, 7/7/09, Mark Campbell-Smith <mcampbellsmith(a)gmail.com> wrote: From: Mark Campbell-Smith <mcampbellsmith(a)gmail.com> Subject: [Freeswitch-users] 2 voicemail questions To: freeswitch-users(a)lists.freeswitch.org Date: Tuesday, July 7, 2009, 8:11 AM Hi! I have 2 questions regarding voicemail ... 1. Can I email the voicemail message to multiple email addresses? If so, what format is this in? <param name="vm-mailto" value="me(a)myemail.com"/> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? Thanks! _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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mod_fifo: Selection of consumers in a certain order.
by Luis F Urrea 07 Jul '09

07 Jul '09

Hi all, We are in the need of a certain application using mod fifo. Basically we are doing the following as described in the wiki: <fifo name="sales_fifo@$${domain}" importance="0"> <member timeout="15" simo="1" lag="5">{call_timeout=30,fifo_member_wait=nowait}user/1009@$${domain}</member> <member timeout="15" simo="1" lag="5">{call_timeout=30,fifo_member_wait=nowait}user/1008@$${domain}</member> </fifo> <extension name="sales_fifo"> <condition field="destination_number" expression="^sales_fifo$"> <action application="answer"/> <action application="set" data="fifo_orbit_exten=1010:45"/> <action application="set" data="fifo_orbit_dialplan=XML"/> <action application="set" data="fifo_orbit_context=default"/> <action application="set" data="fifo_caller_exit_key=2"/> <action application="set" data="fifo_caller_exit_to_orbit=true"/> <action application="fifo" data="sales_fifo@$${domain} in undef $${base_dir}/sounds/music/8000/wooohooo.wav"/> </condition> </extension> Things work fine, but we have noticed that the consumers are selected in kind of a load balancing basis, so that if member 1009 answered the last call then the next call goes to 1008 even if 1009 is available. We would like to know if there is a way to select the next consumer available in a different fashion such as in strict order setting a preference for each member. Say member 1 preference=1 member 2 preference=2 so that only if member 1 is on the phone the call rolls to member 2. Hope it makes sense. All input on how to achieve this is appreciated. Regards,

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2 voicemail questions
by Mark Campbell-Smith 07 Jul '09

07 Jul '09

Hi! I have 2 questions regarding voicemail ... 1. Can I email the voicemail message to multiple email addresses? If so, what format is this in? <param name="vm-mailto" value="me(a)myemail.com"/> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? Thanks!

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Fwd: [UniMRCP] Flite Plugin Available
by Brian West 07 Jul '09

07 Jul '09

Arsen just keeps em coming... Thank you! /b Begin forwarded message: > From: Arsen Chaloyan <achaloyan(a)yahoo.com> > Date: July 7, 2009 10:20:13 AM CDT > To: Brian West <brian(a)freeswitch.org> > Subject: Fw: [UniMRCP] Flite Plugin Available > > > > ----- Forwarded Message ---- > From: Arsen Chaloyan <achaloyan(a)gmail.com> > To: UniMRCP <unimrcp(a)googlegroups.com> > Cc: unimrcp-announcements(a)googlegroups.com > Sent: Tuesday, July 7, 2009 8:16:16 PM > Subject: [UniMRCP] Flite Plugin Available > > I would like to announce the availability of Flite TTS plugin for > UniMRCP server. > Special thanks goes to Garmt, who initially contributed and helped > develop the plugin. > > Currently supported TTS features are as follows: > > English voices: > awb > kal > rms > slt > Methods: > SPEAK > STOP > PAUSE > RESUME > BARGE-IN-OCCURRED > Events: > SPEAK-COMPLETE > Synthesizer Speech Data: > text/plain > > For the instructions on how to build and configure Flite with > UniMRCP refer to > http://code.google.com/p/unimrcp/wiki/FlitePlugin > > Please note, everything is working now, nevertheless this is basic > availability only. > I have mostly tested the integrated solution in the following setup > SIPPhone -> FreeSWITCH/UniMRCPClient -> UniMRCPServer/Flite > > > Feedback is welcome. > Thanks, > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > --~--~---------~--~----~------------~-------~--~----~ > You received this message because you are subscribed to the Google > Groups "UniMRCP" group. > To post to this group, send email to unimrcp(a)googlegroups.com > To unsubscribe from this group, send email to unimrcp+unsubscribe(a)googlegroups.com > For more options, visit this group at http://groups.google.com/group/unimrcp?hl=en > -~----------~----~----~----~------~----~------~--~--- >

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freeswitch & sipnet.ru & caller id
by Alexey Lubimov 07 Jul '09

07 Jul '09

Good day. I have external sip gateway on sipnet.ru Logs from sipnet.ru contain my ip address (193.112.5.111) instead actual number (111(a)193.112.5.111) in field "caller id". Is it possible to set caller id in actual number instead ip address? 07/07/09 14:08 74959345610 193.112.5.111 Russia Moscow 00381982 0:14 0.01750 0.00408

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dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback
by seven 07 Jul '09

07 Jul '09

# play test.wav Input File : 'test.wav' Sample Size : 16-bit (2 bytes) Sample Encoding: signed (2's complement) Channels : 2 Sample Rate : 16000 1) is it the default behavior that uuid_record record with 2 channels 2) is it reasonable that FS can record to 2 channels but cannot playback? 3) do I need to set RECORD_STEREO=false before uuid_record? Thanks for help.

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How to modify my INVITE msg??
by Brad Tuan 07 Jul '09

07 Jul '09

Useless , the dialplan was changed like this: <include> <extension name="1001_trans"> <condition field="destination_number" expression="^1001$"> <action application="bridge" data="sofia/profile/1001123${regex(${sofia_contact( 1001@${domain})}|^[^\@]+(.*)|%1)}"/> </condition> </extension> </include> but when 1003 call 1001 ,the request is still "Request-Line: INVITE sip:1001@192.168.141.182 <sip%3A1001(a)192.168.141.182> SIP/2.0" Dialplan: sofia/internal/1003(a)192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003(a)192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516001$/ break=on-false Dialplan: sofia/internal/1003(a)192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003(a)192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516002$/ break=on-false Dialplan: sofia/internal/1003(a)192.168.141.182 parsing [default->skype_to_1001] continue=false Dialplan: sofia/internal/1003(a)192.168.141.182 Regex (PASS) [skype_to_1001] destination_number(1001) =~ /^1001$/ break=on-false Dialplan: sofia/internal/1003(a)192.168.141.182 Action bridge(sofia/profile/1001123${regex(${sofia_contact( 1001@${domain})}|^[^\@]+(.*)|%1)}) 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1003(a)192.168.141.182) State Change CS_ROUTING -> CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1003(a)192.168.141.182 [BREAK] 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1003(a)192.168.141.182) State ROUTING going to sleep 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1003(a)192.168.141.182) Running State Change CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1003(a)192.168.141.182) State EXECUTE 2009-07-07 10:38:46 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1003(a)192.168.141.182 SOFIA EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1003(a)192.168.141.182Standard EXECUTE EXECUTE sofia/internal/1003(a)192.168.141.182 bridge( sofia/profile/1001123@192.168.141.182:29084;rinstance=b4a8ae8884b9ed6b) 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel() Invalid Profile FS return it is a Invalid Profile....Why??

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How to modify the Subject and Body when sending voicemail??
by Brad Tuan 07 Jul '09

07 Jul '09

As title, How to custom the Subject and Body and ... of the mail ??

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How to modify my INVITE msg??
by Brad Tuan 07 Jul '09

07 Jul '09

Useless , the dialplan was changed like this: <include> <extension name="1001_trans"> <condition field="destination_number" expression="^1001$"> <action application="bridge" data="sofia/profile/1001123${regex(${sofia_contact( 1001@${domain})}|^[^\@]+(.*)|%1)}"/> </condition> </extension> </include> but when 1003 call 1001 ,the request is still "Request-Line: INVITE sip:1001@192.168.141.182 <sip%3A1001(a)192.168.141.182> SIP/2.0" Dialplan: sofia/internal/1003(a)192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003(a)192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516001$/ break=on-false Dialplan: sofia/internal/1003(a)192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003(a)192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516002$/ break=on-false Dialplan: sofia/internal/1003(a)192.168.141.182 parsing [default->skype_to_1001] continue=false Dialplan: sofia/internal/1003(a)192.168.141.182 Regex (PASS) [skype_to_1001] destination_number(1001) =~ /^1001$/ break=on-false Dialplan: sofia/internal/1003(a)192.168.141.182 Action bridge(sofia/profile/1001123${regex(${sofia_contact( 1001@${domain})}|^[^\@]+(.*)|%1)}) 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1003(a)192.168.141.182) State Change CS_ROUTING -> CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1003(a)192.168.141.182 [BREAK] 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1003(a)192.168.141.182) State ROUTING going to sleep 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1003(a)192.168.141.182) Running State Change CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1003(a)192.168.141.182) State EXECUTE 2009-07-07 10:38:46 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1003(a)192.168.141.182 SOFIA EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1003(a)192.168.141.182Standard EXECUTE EXECUTE sofia/internal/1003(a)192.168.141.182 bridge( sofia/profile/1001123@192.168.141.182:29084;rinstance=b4a8ae8884b9ed6b) 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel() Invalid Profile FS return it is a Invalid Profile....Why??

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How to modify my INVITE msg??
by Brad Tuan 07 Jul '09

07 Jul '09

For example, send a "INVITE 1001123(a)xxx.xxx.xxx.xxx" to my FS user 1001 How to do this??

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